cab48c391d
AGC2 component that computes and applies the digital gain. The gain is computed from an estimated speech and noise level. This component decides how fast the gain can change and what it should be. Bug: webrtc:7494 Change-Id: If55b6e5c765f958e433730cd9e3b2b93c14a7910 Reviewed-on: https://webrtc-review.googlesource.com/64985 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22741}
48 lines
1.7 KiB
C++
48 lines
1.7 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/vad/vad_with_level.h"
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namespace webrtc {
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class ApmDataDumper;
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class AdaptiveDigitalGainApplier {
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public:
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explicit AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper);
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// Decide what gain to apply.
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void Process(
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float input_level_dbfs,
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float input_noise_level_dbfs,
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rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results,
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AudioFrameView<float> float_frame);
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private:
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// Keep track of current gain for ramping up and down and
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// logging. This member variable is redundant together with
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// last_gain_db_. Both are kept as an optimization.
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float last_gain_linear_ = 1.f;
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float last_gain_db_ = 0.f;
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// For some combinations of noise and speech probability, increasing
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// the level is not allowed. Since we may get VAD results in bursts,
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// we keep track of this variable until the next VAD results come
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// in.
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bool gain_increase_allowed_ = true;
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ApmDataDumper* apm_data_dumper_ = nullptr;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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